AuthorTopic: Smaart delay finder won't stay put (Read 4206 times)

I recently tried to use my Smaart rig to determine some delays and levels between a line array and some front fills in a ballroom, amongst other tasks. My desk for the week here is a CL5, which is just right for this event.

I decided to try using Dante virtual sound card and the system Dante network to feed Smaart. I sent a direct out of some house music via Dante to Smaart and set that as the reference, and I connected my measurement mics to the desk and sent them as direct outs to Smaart as well as the measurement channels. Transfer function came up seemingly fine, but delay finder would not come up with remotely the same number twice. Even more random than trying to use it on subs - hit it and it says 149ms, hit it again and it says 675ms. I tried the same with the console oscillator, same effect.

Any idea why this would happen? Using Dante is the big thing that was different about this configuration. Besides this it was awesome... I had playback coming from my laptop, and a couple of interesting direct outs feeding dedicated spectrographs in Smaart, and the system TF going all at once with one CAT5 to my laptop. Oh, and CL5 control and Shure UHF-R control on the same network. Maybe it's unrelated, but I don't have any other ideas.

Thanks!

Sorry y'all..... got some questions I've been storing up. Time to tap the knowledge keg.

What Scott said. As Sam Berkow used to say, unless your using heavy metal as your program material it's best to stick with pink noise as a test signal. Even program material that works adequately in a small club will likely not work in a ballroom with a longer RT and greater distances between the mic and the source.

Sam, this may be barking up a tree as i don't know anything about how the dante system network, the CL-5, and the dante virtual card work together,..........but I've found the same delay tracking phenom (and phase jumping) when my reference and measured channels aren't coming from the same digital device in the signal flow.I say digital device because I don't know the right term. What I mean is when digital signals are linked in a stream, there sometimes seem to be issues that drive the smaart delay finder crazy, but that don't cause any audible problems. I'm guessing it's about clock issues between devices, or something else I don't yet understand about digital flow.

It might be as simple as that. I was trying to do it with program material in a very large ballroom.

What has worked best for you guys for doing delays in difficult situations? For starters it sounds like use pink, or pink under music. I guess I do a lot of clubs and etc and I have not had any delay finder issues before, so I have not thought about it.

On the digital question - I've experienced this before when there are two devices with totally independent digital systems. If I recall correctly those were very minor shifts that's caused. I'm no Yamaha networking expert myself... everything is on the Dante network clock, I suppose?

It might be as simple as that. I was trying to do it with program material in a very large ballroom.

What has worked best for you guys for doing delays in difficult situations? For starters it sounds like use pink, or pink under music. I guess I do a lot of clubs and etc and I have not had any delay finder issues before, so I have not thought about it.

On the digital question - I've experienced this before when there are two devices with totally independent digital systems. If I recall correctly those were very minor shifts that's caused. I'm no Yamaha networking expert myself... everything is on the Dante network clock, I suppose?

Sam

I just did a theater rig using DVS to smaart. I switched reference inputs between the desk (also a CL-5) and the DSP with no anomalies.

Everything on one dante network shares a clock from a master device. you can find which device is the master under the clock tab in Dante Controller. But as long as signal is flowing and there appear to be no blips, burps, or chirps, it's probably fine.

I think of the delay finder like early GPS systems, good to +/- 100 meters , alegorically. When it lies to you, 9 times out of 10 it's on account of some local acoustic phenomenon. It's too convenient to not use, but I give it a 3 point cross-check: what does the live IR look like, what is the slope of the phase trace @ nyquist, and is there broadband coherence?

You can change the length and averaging of the delay finder FFT (in V7+ at least). You could try reducing the fft size to window out spurious arrivals and also increase your averaging.

you'll never get a delay measurement on the subs, don't bother. See any number of other threads on this board. Best bet to stick your mic where you want them coincident to the mains, snap your reference delay and a trace to the mains there, and then dial the subs to meet the phase slope of the mains @ xover.

It might be as simple as that. I was trying to do it with program material in a very large ballroom.

What has worked best for you guys for doing delays in difficult situations? For starters it sounds like use pink, or pink under music. I guess I do a lot of clubs and etc and I have not had any delay finder issues before, so I have not thought about it.

Sam

The first time I ran into a major issue finding the delay was on a large outdoor system where the mic was about 100' from the speaker. That's when I learned the pink noise trick and have been using it ever since.

You can still play your normal program material, and add just a little bit of pink to it. You'd be surprised how little pink is required for the system to lock on. The Impulse Response window cleans up a whole lot too when you add pink.

We use dante all the time, with the newer yamaha gear especially. As long as the measurement and reference signals are on the same device, or the devices they are on have sample clock sync - measurements in Smaart will be just fine.

As soon as you don't have sample clock sync, you get exactly what you are describing - and why we don't allow you to reference to the generator or other devices without giving you nasty warnings first.

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I am employed by Rational Acoustics. Besides being a Product Manager for the Smaart Platform, I operate as a fully deployable Smaart Ninja (aka Instructor/System Guy).

We use dante all the time, with the newer yamaha gear especially. As long as the measurement and reference signals are on the same device, or the devices they are on have sample clock sync - measurements in Smaart will be just fine.

As soon as you don't have sample clock sync, you get exactly what you are describing - and why we don't allow you to reference to the generator or other devices without giving you nasty warnings first.

Are you using more than one sound source?If you have L & R mains or Main and front fill, you will get different arrival times and the finder has no idea what to lock on to.Guess why I know.

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Are you using more than one sound source?If you have L & R mains or Main and front fill, you will get different arrival times and the finder has no idea what to lock on to.Guess why I know.

Oh, don't I know it.

The finder will lock onto the highest frequency direct arrival. This could be the FF's even if they are not the loudest system, depending on where you place your microphone/what the system delays are set to/what frequencies are arriving at the measurement probe first.

The best practice is to start with individual systems and work your way from mains to supporting. The goal is to get all of your systems acting as one (system a+b+c+d+etc=A).

You are hitting on why using dual channel measurement for in-situ performance monitoring is not the appropriate tool for the job, especially if you are at a center FOH type position and the whole system is on.

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I am employed by Rational Acoustics. Besides being a Product Manager for the Smaart Platform, I operate as a fully deployable Smaart Ninja (aka Instructor/System Guy).

The finder will lock onto the highest frequency direct arrival. This could be the FF's even if they are not the loudest system, depending on where you place your microphone/what the system delays are set to/what frequencies are arriving at the measurement probe first.

The best practice is to start with individual systems and work your way from mains to supporting. The goal is to get all of your systems acting as one (system a+b+c+d+etc=A).

You are hitting on why using dual channel measurement for in-situ performance monitoring is not the appropriate tool for the job, especially if you are at a center FOH type position and the whole system is on.

+1, Chris.

My workflow is to make initial sound origin measurements from the center line between main hangs. I measure each side side separately (ultimately this can reveal a number of issues with mains), then subs, followed by any fills that may be in coverage of the mains. I'm looking for the "furthest away in time" source. If I'm aligning to backline instruments, whatever is "furthest upstage in time" is aligned to backline, and every other part of the system is delayed back to whatever that loudspeaker subsystem is. Measurement for other uses (voicing, room reflection detection) is done on a one-side-at-a-time basis.

Because of various high pass filters used, sometimes it's not the subsystem one might expect especially if the entire system is not from the same manufacturer or the various subsystems do not use the same DSP settings, HPFs, etc. Nothing like discovering that the attempt to delay subwoofers to mains doesn't work because the main PA is "ahead" of the subs... and I've seen this happen with BEs attempting hasty alignments.

To channel an Ivanism - "It Depends."

So the take-away is that any high pass filters used should be in place *before* making impulse measurements (on which the Smaart delay finder is based) and any voicing EQ should also be roughed in, too.

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